曙海教育集团
全国报名免费热线:4008699035 微信:shuhaipeixun
或15921673576(微信同号) QQ:1299983702
首页 课程表 在线聊 报名 讲师 品牌 QQ聊 活动 就业
 
Tinc VPN培训
 
   班级人数--热线:4008699035 手机:15921673576( 微信同号)
      增加互动环节, 保障培训效果,坚持小班授课,每个班级的人数限3到5人,超过限定人数,安排到下一期进行学习。
   授课地点及时间
上课地点:【上海】:同济大学(沪西)/新城金郡商务楼(11号线白银路站) 【深圳分部】:电影大厦(地铁一号线大剧院站)/深圳大学成教院 【北京分部】:北京中山学院/福鑫大楼 【南京分部】:金港大厦(和燕路) 【武汉分部】:佳源大厦(高新二路) 【成都分部】:领馆区1号(中和大道) 【广州分部】:广粮大厦 【西安分部】:协同大厦 【沈阳分部】:沈阳理工大学/六宅臻品 【郑州分部】:郑州大学/锦华大厦 【石家庄分部】:河北科技大学/瑞景大厦
开班时间(连续班/晚班/周末班):即将开课,详情请咨询客服!
   课时
     ◆资深工程师授课
        
        ☆注重质量 ☆边讲边练

        ☆若学员成绩达到合格及以上水平,将获得免费推荐工作的机会
        ★查看实验设备详情,请点击此处★
   质量以及保障

      ☆ 1、如有部分内容理解不透或消化不好,可免费在以后培训班中重听;
      ☆ 2、在课程结束之后,授课老师会留给学员手机和E-mail,免费提供半年的课程技术支持,以便保证培训后的继续消化;
      ☆3、合格的学员可享受免费推荐就业机会。
      ☆4、合格学员免费颁发相关工程师等资格证书,提升您的职业资质。

课程大纲
 

Part I: Introduction

Introduction
History and motivation
Types of VoIP and its evolution
SIP – main concepts
SIP standardization (RFC 3261 and other relevant standards)
Architecture
UA – User Agent
Predefined servers: Registrar, Location, Proxy and Redirect
Application servers
Identification and addressing
SIP trapezoid
Servers and their operation
Registration
SIP server in Proxy and Redirect modes
Stateless and stateful Proxy servers
Location server
SRV records and DNS
uri/url/urn, ENUM and NAPTR records
SIP signalling messages (including Instant Messaging & Presence – IMP extensions)
Message structure
Requests
Responses
Example of a call
Headers and parameters
IMP models
SDP (Session Description Protocol)
Description of media
Standard list of codecs
Session negotiation rules
Call flows – SIP signalling
SIP session – main RFC 3261 example
Sample call scenarios
Conferencing and IP PBX
Changing media during a session
Using IMP
Routing of SIP requests and responses
VIA header
ROUTE and RECORD-ROUTE headers
SIP-PSTN interworking
SIP-T and SIP-I
SIP early media and SIP trunking
SIP-PSTN signalling
SIP – security problems
Secure SIP, Secure RTP and Secure RTCP
Typical implementations of Secure SIP
Practical problems and perspectives
NAT and firewall traversal
QoS
SIP and SDP in 3GPP IMS architecture
Wrap-up and discussion
Part II: Hands on

SIP in LAN environment: XLite SIP UA + Asterisk
Creating Asterisk accounts with a simple dial plan
Configuration of XLite SIP UA (dtmf, codecs, nat, rtp, timer, register) and SIP phones (Polycom, Gigaset, Yealink, Linphone)
Registration, initiating and receiving calls
P2P calls with Linphone
Analyzing of SIP signalling using Wireshark
Configuration of a server
Registration of SIP signalling and RTP media streams
SIP packet analysis. Retrieval of a specific call
Voice quality problems. Jitter buffer. Retrieval of DTMF signalling (RFC 2833, INFO). Codec and DTMF troubleshooting (transcoding, GSM codec failure, DTMF tone duplication)
VoIP monitor
SDP, Instant Messaging and Presence (IM&P)
SDP parameters and attributes
SUBSCRIBE, PUBLISH and MESSAGE SIP methods
Practising IM&P with XLite and Linphone
SIP call flows
SIP Registration with DNS
SIP SRV record
SIP phone registration using DNS-SRV
Call Flows with DNS
Analysing SIP call signalling using Wireshark
Troubleshooting – DNS timeout, latency
SIP trunks
Establishing a test SIP trunk
Troubleshooting (DOS, DDOS, fraud, cps)
SIP security issues
SIP security with IPSec
Security with Secure SIP
IP telephony – risk of frauds
Preventing DDOS and other types of attacks
Launching SIP based VoIP services
Configuration of a switch
SIP client configuration and registration
Software
Asterisk PBX / Freeswitch softswitch / Cisco Call Manager
Linux CentOS
TDM2IP drivers
Softphones (XLite, Linphone)
Hardware
Server
TDM2IP card/gateway
Hardphone (Polycom, Gigaset, Yealink)
Softphone/Hardphone
Configuration
Codecs
User/Password/SIP Server/Proxy/Ports
Operation and signalling for:
3-Way Calling
Call Forwarding
Attendant Call Transfer
MWI, BLF
Yealink autoprovisioning
Vendor dependent constraints
SIP & Network Adress Translation (NAT) problems
Type and structure of NATs
STUN (Simple Traversal of UDP Through NATs)
Quality of VoIP calls – troubleshooting
Call connected – missing media
Key QoS factors
Delay, jitter, play buffer size
VoIP quality metrics
RTCP – delay and jitter
MOS according to ITU-T G.107 E-model
VoIP quality monitoring tools (Voipmonitor)
Cloud based IP telephony
Wrap up and addressing SIP and VoIP related issues submitted by participants

 
 
  备案号:沪ICP备08026168号 .(2014年7月11)...................
友情链接:Cadence培训 ICEPAK培训 PCB设计培训 adams培训 fluent培训系列课程 培训机构课程短期培训系列课程培训机构 长期课程列表实践课程高级课程学校培训机构周末班培训 南京 NS3培训 OpenGL培训 FPGA培训 PCIE培训 MTK培训 Cortex训 Arduino培训 单片机培训 EMC培训 信号完整性培训 电源设计培训 电机控制培训 LabVIEW培训 OPENCV培训 集成电路培训 UVM验证培训 VxWorks培训 CST培训 PLC培训 Python培训 ANSYS培训 VB语言培训 HFSS培训 SAS培训 Ansys培训 短期培训系列课程培训机构 长期课程列表实践课程高级课程学校培训机构周末班 曙海 教育 企业 学院 培训课程 系列班 长期课程列表实践课程高级课程学校培训机构周末班 短期培训系列课程培训机构 曙海教育企业学院培训课程 系列班